In traditional circuit-switched networks, such as the Public Switched Telephony System (“PSTN”), each user endpoint is connected to at most one switching system. In a business enterprise, a business telephone is connected to a single Private Branch Exchange (“PBX”). A PBX is an intelligent switching point within a circuit-switched network that is responsible for routing calls to a plurality of internal nodes or public destinations via a single PSTN switching system.
Newer telephony networks that employ packet-switching technologies are growing in popularity. In particular, packet-switched telephony networks that use the Internet Protocol (IP) as a network protocol for transmitting and receiving voice data are becoming more prevalent. These so-called Internet Telephony networks have the potential to offer new features and services that are currently unavailable to subscriber of circuit-switched telephony networks. Conceptually, IT Networks differ from the PSTN systems in that they generally transmit voice data exchanged between two subscriber endpoints, according to an IP format. More specifically, they encapsulate voice data into data packets, which are transmitted according to an IP format in the same manner as textual data is transmitted from one computer to another via the internet.
The Session Initiation Protocol (SIP) is one of several protocols that may be used with the Internet Protocol to support Internet Telephony applications. The SIP specification is defined in the Internet Engineering Task Force (IETF) Request for Comments (RFC) 3261, dated June 2002; the disclosure of which is incorporated herein by reference in its entirety. SIP is an application-layer control protocol for creating, modifying, and terminating sessions between networked endpoints, which are referred to as SIP Enabled Devices, User Agents or simply SIP endpoints.
As discussed above, SIP Enabled Devices implement a network communication protocol, wherein a communication session is established for two endpoints to transmit and receive data. As such, each SIP Device in a SIP network is assigned a unique SIP address or terminal name, which is defined in a SIP Universal Resource Identifier (URI). The format of a SIP URI is similar to that of an email address, which typically includes a user name “at” a domain name, for example “sip:alice@siemens.com.” SIP URI data is placed into header fields of SIP messages, for example to identify a sender and a receiver of the SIP message. For secure communications, the SIP Specification also defines a SIPS URI, for example “sips:alice@siemens.com.” Accordingly, when a SIPS URI is used the SIP Enabled Device associated with the SIP URI may implement an encryption protocol for transmitting data in a secure communication session. It should be noted that the SIPS URI protocol may be used interchangeably with SIP URI protocol, in the examples that follow.
The SIP specification defines several types of communication resources implemented for establishing and maintaining SIP based communication sessions among SIP Enabled Devices, which include SIP Registrars, SIP Redirect Servers, and SIP Proxies. These SIP communication resources are responsible for sending, receiving, routing, and relaying SIP messages among SIP Enabled Devices.
SIP Proxy Servers perform a variety of functions in SIP networks, such as coordinating data routing for SIP session requests to a particular SIP network subscriber current location; authenticating and authorizing SIP network users for particular services; implementing call-routing policies, as well as providing additional functionality to SIP network users. A conventional SIP endpoint is capable of identifying and displaying the terminal name and/or terminal number (Host Address/user information) of a subscriber's endpoint/telephone based on SIP URI address information. More specifically, a SIP endpoint is configured to process the SIP URI address information to identify an incoming telephone/endpoint and register the endpoint to a proxy/back-to-back user agent. By way of example only, analysis of the SIP URI data may determine user information such as the terminal number associated with an endpoint, such as extension 4444 assoicated with a terminal. However, the information stored within a conventional SIP Enabled Device and displayed to a user are parameters used for data routing. They do not necessarily correspond to a Direct Inward Dial (DID)/public number (e.g., the main number for a business, a DID number) or a non-DID/private number (e.g., one-to-six digits, often displayed as an extension number which cannot be accessed directly by a public terminal. Conventional SIP Enabled Devices may also display the host address of the endpoint, which depending on the implementation may be deceptively similar to a viable telephone number (e.g., 561-55X-1234 or subscriberA@company.com). However, this data, the terminal number and terminal name, from the SIP URI are used for data routing by the SIP Enabled Devices involved in a communication session.
The SIP URI data is stored in a SIP Enabled Device and can be used as part of information that is displayed to a user. For example, in a conventional SIP Enabled Device, the SIP Enabled Application, a controller program stored in a device's memory and executed by a SIP Enabled Device's processor, may parse the stored SIP URI associated with the device and display a terminal name and/or the terminal number. However, as discussed above this information is used in data routing between communication nodes in a network and may or may not correspond to a public number (DID) or a private number (non-DID).